Delta-sigma modulators (noise shapers) are particularly useful in digital to analog and analog to digital converters (DACs and ADCs). Using oversampling, a delta-sigma modulator spreads quantization noise power across the oversampling frequency band, which is typically much greater than the input signal bandwidth. Additionally, a delta sigma modulator performs noise shaping by acting as a lowpass filter to the input signal and a highpass filter to the noise; most of the quantization noise power is thereby shifted out of the signal band. U.S. Pat. No. 5,815,102 entitled “Delta Sigma PWM DAC to Reduce Switching” to John Melanson granted Sep. 29, 1998 and U.S. Pat. No. 6,150,969 entitled “Correction of Nonlinear Output Distortion in a Delta Sigma DAC” to John Melanson granted on Nov. 21, 2000 and U.S. Pat. No. 6,480,129 entitled “Methods and Apparatus for Correction of Higher Order Delta Sigma Converters” to John Melanson granted on Nov. 12, 2002 disclose exemplary ways for implementing the modulator and are incorporated herein by reference.
In addition to data conversion applications, delta-sigma noise shapers are increasingly utilized in the design of digital amplifiers. In one particular technique, a digital delta-sigma noise shaper provides a noise shaped (quantized) digital data stream to a pulse width (duty cycle) modulated (PWM) stream, which in turn drives a linear amplifier output stage and associated load. This technique is generally described in U.S. Pat. No. 5,784,017, entitled “Analogue and Digital Convertors Using Pulse Edge Modulators with Non-linearity Error Correction”, granted Jul. 21, 1998, and U.S. Pat. No. 5,548,286, entitled “Analogue and Digital Convertor Using Pulse Edge Modulators with Non-linearity Error Correction”, granted Aug. 20, 1996, both to Craven, U.S. Pat. No. 5,815,102, entitled “Delta Sigma PWM DAC to Reduce Switching”, granted Sep. 29, 1998, U.S. patent application Ser. No. 09/163,235 to Melanson, and International Patent Application No. PCT/DK97/00133 by Risbo, all of which are incorporated herein by reference.
Another example is disclosed in U.S. Pat. No. 6,480,129 entitled “Methods and Apparatus for Correction of Higher Order Delta Sigma Converters” to John Melanson granted on Nov. 12, 2002, and incorporated herein by reference, which teaches reducing EMI through dithering. Additionally, these techniques only marginally reduce the audio signal quality.
U.S. Pat. No. 6,384,761 to Melanson, issued May 7, 2002 and incorporated herein by reference discloses a Second and higher order dynamic element matching in multibit digital to analog and digital to analog data converters A multibit data converter has an output parallel unit element converter fed by a multibit signal, and noise shaping dynamic element matching (DEM) apparatus for selectively activating units in the converter. The DEM apparatus includes a plurality of noise shaping components. Each component has as an input one signal to the converter, and each includes a first integrator having as its input the input to the component, and a second integrator having as its input the output of the first integrator, and forms one or more component outputs. A signal in the second integrator is clipped. A vector quantizer orders the component outputs and activates converter elements according to the ordering.
Sample-rate conversion (SRC) refers to the process of translating a sequence of discrete data samples with a certain sampling rate into another sequence with a different sampling rate, while preserving the information contained in the original sequence. Sample rate conversion can be a relatively simple technique where the two sample rates are related by a fairly small integer ratio. For example, if the ratio of sample rates is 4:5, the input signal only needs to be upsampled by a factor of four and then, after appropriate filtering, down-sampled by a factor of five.
FIG. 1 illustrates an example of such a Prior Art sample rate conversion technique. In this Prior Art embodiment, input data may be upsampled by a factor of M in upsampler 110, to produce an upsampled data stream. Upsampling may comprise the process of creating new intermediate samples by inserting samples of value zero as intermediate values. Upsampled data may be filtered in low pass filter 120, which may comprise a digital filter. The resultant data may be downsampled by a factor of N in downsampler 130 which may decimate unwanted data samples. Downsampling may comprise decimating (eliminating) intermediate sample values to create a new sample set at the desired output sample rate.
Low pass filter 120 may be used to eliminate spectral “images” produced when upsampling. Such images are illustrated in FIG. 2. Referring to FIG. 2, the sample data at rate fs is illustrated, along with image data produced during upsampling at sample rates 2fs and 3fs. FIG. 3 illustrates how filtering may be used to eliminate undesirable images while preserving desirable data, as illustrated in FIG. 4 to produce sample rate converted data as illustrated in FIG. 5.
In the example given above for a ratio of sample rates of 4:5, M would equal 4 and N would equal 5. Such a Prior Art sample rate conversion technique may be suitable for relatively simple rate conversions. However, not all scenarios where sample rate conversion is desired have such simple sample rate ratios. For example, when converting by more drastic ratios (e.g., 124:359) the amount of processing required to upsample and then downsample may be prohibitive.
As compared to the simpler scenario where the sampling rates are related by a relatively simple ratio, the term “asynchronous sample-rate conversion” is often used to signify situations where the ratios are rather ill-defined numbers such that their least common multiple is a very large number. Asynchronous sample rate conversion is particularly applicable in cases where data is at the “same” sampling rate, but created with a different clock crystal, and thus at a slightly different sampling frequency. Thus, one set of data might have a sample rate of a “fast” 48 KHz and another might have a sample rate of a “slow” 48 KHz. Such ratios like this that are close to 1:1, but not exact, fall into the category of “asynchronous sample-rate conversion.” Asynchronous sample-rate converters are commonly used when bridging two systems that are on different sampling rates and/or master operating clocks.
Additionally, in the field of audio, there are various standards that utilize different sampling rates such as 32 KHz, 44.1 KHz, 48 KHz, and so on, and sample rate converters are commonly used to interconnect different systems with the different standards.
There have been proposed different methods of sample rate conversion. FIGS. 6-10 illustrate the operation of one such Prior Art sample rate conversion scheme. As illustrated in FIG. 6, such a sample rate converter interpolates (i.e., upsamples) a source signal Fsi 1205 in xU interpolator 1210 to a higher sample rate (either physically or conceptually) so that it can represent the original analog signal within a prescribed error bound. FIG. 7 represents the frequency distribution of the input signal Fsi at the input sample rate. A reconstruction filter 1220 eliminates artifacts present in the up-sampled signal. FIG. 8 illustrates the reconstruction filter response as applied to the interpolated signal. The upsampled signal is then re-sampled in downsampler 1230 at a certain multiple D of the desired rate Fso. The re sampled sequence is then further decimated down by decimator 1240. FIG. 9 illustrates the action of the decimation filter. The resultant data Fso at the output sample rate is output 1250 as illustrated in FIG. 10.